[Serusers] loop call with asterisk

[Richard]

Hi,

I am trying to use * as a PSTN gateway. When a call comes in from PSTN, it
is forwarded to ser. If for some reason (e.g. call being forwarded back to
PSTN), ser will just do a record route and send it back to *. When * gets
this call, it thinks it is a loop and drop the call. I know that this
question has been raised before. I'd like to see, just from sip respective,
what's the theoretical way to solve it? Since all major fields are the same,
e.g. fromuri, touri, cseq, callid, what is the right way to detect loop in
SIP in this case? Btw, cisco router doesn't have this problem.

Thanks,
Richard